[asterisk-users] need help on setup rtp directly between 2 sip clients

[haloha]

Hi Alyed

xilte softphone work perfectly on other sip server(opensips server)

Don't remember the exact syntax but guess it's something like
sip:username at the.phones.ip:
>
> 5060


>>>you mean i config the extension.conf look like exten =>
1000,1,Dial(SIP/1000 at ip address:5060), is it right?

the problem i got here is the asterisk server to stay middle of  media
first, then redirect the media later, how to fix it,asterisk no need stay in
middle of media because all devices are in the same LAN

is there another hint

Thank you


On Fri, Mar 26, 2010 at 11:56 PM, Alyed <alyed at vivoxie.com> wrote:

> I guess to do what you want you need to dial directly between the phones.
> Can't do it with xlite but you can with SJphones
>
> Don't remember the exact syntax but guess it's something like
> sip:username at the.phones.ip:5060
>
> Alyed
>
>
>
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